1. Field of the Invention
This invention relates to sound or speech signal processors in which frequency transformation of a recorded sound signal is provided for the purpose of restoring the original frequency components of the signal. More particularly, the invention pertains to improved speech signal processors of the above-stated character in which a comb filter is connected at the input or output side of the signal processor to minimize noise components in the output signal.
2. Description of the Prior Art
Speech signal transformation processors which permit a speech signal input to be compressed or expanded in time so as to audibly reproduce the signal at its normal frequency spectrum are well known. Typical arrangements are exemplified by U.S. Pat. Nos. 1,671,151; 2,352,023; and 3,480,737. The systems disclosed in these patents, when used to reduce the frequency of a speech signal, while compressing the time in which a given segment of speech is reproduced, inevitably involve discarding a portion of the original speech wave. The ratio of the speech signal discarded to that which is retained is directly related to the compression ratio and the discard loss is inhererently and fundamentally related to this process of reducing the frequency and compressing the time for the processing of a given passage of speech. Since the portion of the speech which is reproduced alternates with portions which are discarded, the problem of merging to reproduce sections in continuous time slots presents some problem and various solutions have been offered.
U.S. Pat. No. 3,786,195, issued to Schiffman, suggests a signal controlled delay line disposed directly in a sound signal channel between the signal source and sound reproducer, which delay line is repeatedly sequenced between maximum and minimum delay values to modify the frequency-time characteristic of the sound reproduced from the original signal. This sytem provides signal processing at the point of juncture of two reproduced speech segments to suppress distracting noise components and also to avoid the introduction of false cues which could modify the information conveyed in the subsequent speech segment. To this end, the transition between successive reproduced speech samples are modified by simple transfer function selection or control, or the transition is eased by the introduction of synthetic or speech-derived signal portions to approximate a smooth transition within a time interval which does not lose actual cues and under such conditions that do not introduce false cues. This system has proved to be effective, but suffers from the disadvantage that the circuitry required is considerably complex.